Adding a sip trunk profile to your telephone system to allow calling in and out via Paytia
When connecting your telephone system to the Paytia SIP as a service product there will be several settings you will need to setup on your SIP trunk connection.
- IP Addresses
- Firewall ports
- SIP URL header format
- Telephone number presented
IP address. Paytia works on a trusted IP address model. You will need to provide Paytia Support with the IP address your telephone system presents when making outbound calls.
You should configure your telephone system to connect to Paytia’s SIP IP endpoint, 18.104.22.168. This is an elastic IP address, so it stretches across the Paytia load balanced services.
If you are an existing Paytia user and need your telephone system registered please complete this form IP registration form
SIP firewall ports. Paytia will send and receive calls using UDP port 5060 for SIP or TCP 5061 for TLS SIPS.
Media firewall ports. Paytia will react to the media port range you set. Or we can agree an accepted range e.g. UDP 10,000 – 20,000
Codec – Paytia work with G.711 A-law, G.711 U-law and G.729 in this order. Codecs control how the audio of your telephone calls are compressed and sent. The key point here is the Paytia side and your PABX side must match.
SIP URL header
SIP header URL – Paytia is expecting a call to its platform from your PABX in the format
sip:email@example.com or sip:1-999-123-4567@Your_Internet_Public_IP_Address
Presentation telephone number (Trunk CLI)
The telephone number your telephone system presents when making outbound calls should match the telephone number you setup on your Paytia account. We will use this number as an extra match value for accepting calls from your telephone system.